Adds a VoIP.ms SIP trunk on top of asterisk-digital-ocean: IP-authenticated
trunk (no password stored), NANP-only outbound dialplan, a global 3-call
concurrent cap via GROUP()/GROUP_COUNT(), and inbound routing to one
extension. Config lives in its own include files rather than being
appended directly to pjsip.conf/extensions.conf, since Easy Asterisk fully
regenerates both from its own internal state — the includes are patched
into the vendor's generator functions so they survive that regeneration.
Wires the new service into setup.sh's is_installed() and README's services
table, and updates docs/pstn-calling-voipms-plan.md to reflect what's now
implemented vs. still open (spend/volume alerting, live-account
verification).
Inbound (DID) is now decided as wanted, not outbound-only. Adds a cost
estimate table for 100 min/month each direction, and clarifies that
NANP-only restriction bounds cost-per-minute but not burn speed, so the
concurrent-call cap and spend alert are required before funding a live
trunk, not optional hardening.
Captures the toll-fraud / prepaid-cap research and decisions from a design
discussion (provider: VoIP.ms, US-only calling for now) so implementation
can be picked up in a future session without re-deriving the background.
Nothing implemented yet.
Claude